swr: Add SOX resampler support

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

configure

@@ -213,6 +213,7 @@ External library support:
   --enable-libpulse        enable Pulseaudio input via libpulse [no]
   --enable-librtmp         enable RTMP[E] support via librtmp [no]
   --enable-libschroedinger enable Dirac de/encoding via libschroedinger [no]
+  --enable-libsoxr         enable Include libsoxr resampling [no]
   --enable-libspeex        enable Speex de/encoding via libspeex [no]
   --enable-libstagefright-h264  enable H.264 decoding via libstagefright [no]
   --enable-libtheora       enable Theora encoding via libtheora [no]
@@ -1173,6 +1174,7 @@ CONFIG_LIST="
     libpulse
     librtmp
     libschroedinger
+    libsoxr
     libspeex
     libstagefright_h264
     libtheora
@@ -3839,6 +3841,7 @@ enabled libopus    && require_pkg_config opus opus_multistream.h opus_multistrea
 enabled libpulse && require_pkg_config libpulse-simple pulse/simple.h pa_simple_new
 enabled librtmp    && require_pkg_config librtmp librtmp/rtmp.h RTMP_Socket
 enabled libschroedinger && require_pkg_config schroedinger-1.0 schroedinger/schro.h schro_init
+enabled libsoxr    && require  libsoxr soxr.h soxr_create -lsoxr
 enabled libspeex   && require  libspeex speex/speex.h speex_decoder_init -lspeex
 enabled libstagefright_h264  && require_cpp libstagefright_h264 "binder/ProcessState.h media/stagefright/MetaData.h
     media/stagefright/MediaBufferGroup.h media/stagefright/MediaDebug.h media/stagefright/MediaDefs.h
@@ -4254,6 +4257,7 @@ echo "libopus enabled           ${libopus-no}"
 echo "libpulse enabled          ${libpulse-no}"
 echo "librtmp enabled           ${librtmp-no}"
 echo "libschroedinger enabled   ${libschroedinger-no}"
+echo "libsoxr enabled           ${libsoxr-no}"
 echo "libspeex enabled          ${libspeex-no}"
 echo "libstagefright-h264 enabled    ${libstagefright_h264-no}"
 echo "libtheora enabled         ${libtheora-no}"

libswresample/Makefile

@@ -13,4 +13,6 @@ OBJS = audioconvert.o                        \
        resample.o                            \
        swresample.o                          \
 
+OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o
+
 TESTPROGS = swresample

libswresample/resample.c

@@ -196,7 +196,8 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap
 }
 
 static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
-                                    double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta){
+                                    double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,
+                                    double precision, int cheby){
     double cutoff = cutoff0? cutoff0 : 0.8;
     double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
     int phase_count= 1<<phase_shift;

libswresample/soxr_resample.c

+/*
+ * audio resampling with soxr
+ * Copyright (c) 2012 Rob Sykes <aquegg@yahoo.co.uk>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio resampling with soxr
+ */
+
+#include "libavutil/log.h"
+#include "swresample_internal.h"
+
+#include <soxr.h>
+
+static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
+        double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){
+    soxr_error_t error;
+
+    soxr_datatype_t type =
+        format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S :
+        format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I :
+        format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S :
+        format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I :
+        format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S :
+        format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I :
+        format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S :
+        format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1;
+
+    soxr_io_spec_t io_spec = soxr_io_spec(type, type);
+
+    soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby);
+    q_spec.bits = linear? 0 : precision;
+    q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;
+
+    soxr_delete((soxr_t)c);
+    c = (struct ResampleContext *)
+        soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
+    if (!c)
+        av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
+    return c;
+}
+
+static void destroy(struct ResampleContext * *c){
+    soxr_delete((soxr_t)*c);
+    *c = NULL;
+}
+
+static int flush(struct SwrContext *s){
+    soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
+    return 0;
+}
+
+static int process(
+        struct ResampleContext * c, AudioData *dst, int dst_size,
+        AudioData *src, int src_size, int *consumed){
+    size_t idone, odone;
+    soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
+    error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
+            &idone, dst->ch, (size_t)dst_size, &odone);
+    *consumed = (int)idone;
+    return error? -1 : odone;
+}
+
+static int64_t get_delay(struct SwrContext *s, int64_t base){
+    double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate;
+    return (int64_t)(delay_s * base + .5);
+}
+
+struct Resampler const soxr_resampler={
+    create, destroy, process, flush, NULL /* set_compensation */, get_delay,
+};
+

libswresample/swresample.c

@@ -86,6 +86,9 @@ static const AVOption options[]={
 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},
+{"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},
+{"precision"            , "set resampling precision"    , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
+{"cheby"                , "enable Chebyshev passband"   , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
@@ -262,6 +265,10 @@ av_cold int swr_init(struct SwrContext *s){
     }
 
     switch(s->engine){
+#if CONFIG_LIBSOXR
+        extern struct Resampler const soxr_resampler;
+        case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
+#endif
         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
         default:
             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
@@ -272,7 +279,7 @@ av_cold int swr_init(struct SwrContext *s){
     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 
     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
-        s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta);
+        s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby);
     }else
         s->resampler->free(&s->resample);
     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
@@ -491,7 +498,7 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count,
             }
         }
 
-        if(in_count && !s->in_buffer_count){
+        if((s->flushed || in_count) && !s->in_buffer_count){
             s->in_buffer_index=0;
             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
             out_count -= ret;

libswresample/swresample.h

@@ -117,6 +117,7 @@ enum SwrDitherType {
 /** Resampling Engines */
 enum SwrEngine {
     SWR_ENGINE_SWR,             /**< SW Resampler */
+    SWR_ENGINE_SOXR,            /**< SoX Resampler */
     SWR_ENGINE_NB,              ///< not part of API/ABI
 };
 

libswresample/swresample_internal.h

@@ -77,6 +77,8 @@ struct SwrContext {
     double cutoff;                                  /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
     enum SwrFilterType filter_type;                 /**< resampling filter type */
     int kaiser_beta;                                /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
+    double precision;                               /**< resampling precision (in bits) */
+    int cheby;                                      /**< if 1 then the resampling FIR filter will be configured for maximal passband flatness */
 
     float min_compensation;                         ///< minimum below which no compensation will happen
     float min_hard_compensation;                    ///< minimum below which no silence inject / sample drop will happen
@@ -125,7 +127,7 @@ struct SwrContext {
 };
 
 typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
-                                    double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta);
+                                    double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
 typedef void    (* resample_free_func)(struct ResampleContext **c);
 typedef int     (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
 typedef int     (* resample_flush_func)(struct SwrContext *c);